Re: [Asterisk-Users] GSM Gateway / Terminal for sale

2005-11-17 Thread danny zak
send one over

i'll wire it

On Thu, Nov 17, 2005 at 12:50:29PM -, Sam Tam wrote:
 
 Single port GSM Gateway support 900 / 1800 GSM mode with external antenna.
 
 Brand new unit and just been out for testing only.
 Extremely easy to setup and can be used out of the box without any
 configuration. So should be good alternatively of phonecell or nokia pbx
 etc..
 
 Units are located in UK and £60 GBP per unit excluding shipping.
 
 I have only got 2-3 lefts as they are excess stocks hence to avoid
 disappointment 
 
 Working mode: GSM 900 MHz or GSM 1800MHz double frequency 
 Peak power: 2 W
 Power consume: static state 25mA, launch 600mA
 Sencitivity:-104dB
 Inner pressure :DC 12V/1.5A
 Condition temperature:0C~+40C
 Working humidity:45%-90%
 Atmosphere pressure:86~106Pka
 Circumstance noise:60 dB
 Wireless decibel :3.5dB or 12dB
 AC power:220V ac+-10%,frequency 47-54Hz;110Vac/60Hz(optional)
 Power port: China, USA, UK, (by customer ‘s optional)
 Connection means:RJ-11 telephone line plug
 Antenna connection: SMA antenna tie-in, N type port(optional).TNC
 port(optional)
 
 
 Sam
 
 
 
 
 
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[Asterisk-Users] transfer call ?

2004-10-14 Thread Danny Zak
Hello Asterisk dudes and dudetes;

i got -what seems to me a simple a fair question- question about
transferring a call during a conversation.

Like in the following example;

CAPI takes call; sends it out to SIP/phone1

the operator of phone1 picks up; talks to the caller; and after a
while he decides to dispatch/transfer/send out the call the a other
operator that is on phone2.

WHAT do i need to do on my phone to get it transfered ?

i'm using : Welltech 1501 and a standard analog phone (that has got a
flash button)

if i press flash and dial the extensions; nothing happens; i though
normally that by pressing flash you should arrive in a other mode that
permits you to send dtmf in the background.. but in my situation it is
just being sent over to the caller.

ALL my phones and sip devices are configured as outband dtmf; also my
CAPI.

any luck ?

-- 
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 Danny  mailto:[EMAIL PROTECTED]

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Re: [Asterisk-Users] Re: Re[2]: cisco ip 7905 legal ..

2004-10-11 Thread Danny Zak
Hello Pavel,

well .. any GOOD propisition for the same or lower price would be nice

IP300 and ip500 are more expensif than this one



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Monday, October 11, 2004, 12:22:45 PM, you wrote:

PJ so, better is to look to another phone, than surcharge cisco ;-)
PJ PJ




PJ - Original Message - 
PJ From: AST 386sx 
PJ Newsgroups: gmane.comp.telephony.pbx.asterisk.user
PJ Sent: Monday, October 11, 2004 2:00 AM
PJ Subject: Re: Re[2]: cisco ip 7905 legal ..


PJ If you are going to buy it new.  It should not be a problem at all.  Just
PJ order the SIP software with the phone.

PJ Your order should be somthing like this.
PJ CP-7905G - Cisco IP Phone 7905G, Global
PJ SW-SMH-UL-7905 - SIP or H.323 license for single 7905 IP phone
PJ CON-SNT-CP7905 - 8x5xNBD Svc, Cisco IP Phone 7905

PJ Not the following software(s).
PJ SW-CCME-UL-7905 - Cisco CallManager Express License For Single 7905 IP Phone
PJ SW-CCM-UL-7905 - CallManager Unit license for single 7905 IP phone

PJ Both softwares should be same price.  I don't think you can change the
PJ software license or type of software even you hav smartnet.

PJ --ast386--


PJ - Original Message - 
PJ From: Danny Zak [EMAIL PROTECTED]
PJ To: Kannaiyan Natesan [EMAIL PROTECTED]
PJ Cc: Asterisk Users Mailing List - Non-Commercial Discussion 
PJ [EMAIL PROTECTED]
PJ Sent: October 10, 2004 3:32 PM
PJ Subject: Re[2]: [Asterisk-Users] cisco ip 7905 legal ..


 Hello Kannaiyan,

 i need to know the correct procedure; otherwise i will bringing my
 customers in danger and that is not what i want.

 i know you can buy the 7905 WITHOUT the callmanager license.. if i
 load the sip image in it; will that be ok ?


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[Asterisk-Users] cisco ip 7905 legal ..

2004-10-10 Thread Danny Zak
Hello Asterisk,

  can somebody tell me what to purchase EXACTLY for not violating the
  licensee agreements of this phone in according to use it properly
  with *

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Re[2]: [Asterisk-Users] cisco ip 7905 legal ..

2004-10-10 Thread Danny Zak
Hello Kannaiyan,

i need to know the correct procedure; otherwise i will bringing my
customers in danger and that is not what i want.

i know you can buy the 7905 WITHOUT the callmanager license.. if i
load the sip image in it; will that be ok ?


-- 
Best regards,
 Dannymailto:[EMAIL PROTECTED]

belGOnet.com  a  Euro-pictures  division - internet solutions
place princesse elisabeth 9/11   -   1030 Brussels  - Belgium
Tel : +32-(0)2-215.67.65  -  Fax : +32-(0)2-215.66.65

domains - hosting - hardware - VoiP - consultancy - backuping
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Sunday, October 10, 2004, 11:16:11 PM, you wrote:

KN oo. Is there is anything such so.
KN When you buy it for you, does that device not belonged to you?

KN I don't think cisco can catch you if you load linux on that and use that as
KN a computer.

KN -Kannaiyan


KN - Original Message - 
KN From: Danny Zak [EMAIL PROTECTED]
KN To: Asterisk Users Mailing List - Non-Commercial Discussion 
KN [EMAIL PROTECTED]
KN Sent: Sunday, October 10, 2004 10:10 PM
KN Subject: [Asterisk-Users] cisco ip 7905 legal ..


 Hello Asterisk,

  can somebody tell me what to purchase EXACTLY for not violating the
  licensee agreements of this phone in according to use it properly
  with *

 -- 
 Best regards,
 Danny  mailto:[EMAIL PROTECTED]

 belGOnet.com  a  Euro-pictures  division - internet solutions
 place princesse elisabeth 9/11   -   1030 Brussels  - Belgium
 Tel : +32-(0)2-215.67.65  -  Fax : +32-(0)2-215.66.65

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Re[2]: [Asterisk-Users] Asterisk - WellGate 3502a : ulaw/alaw only?

2004-09-26 Thread Danny Zak
Hello Dinesh,

my welltech does support g729 g723.1 and g711u and a.

It works like a charm here.

BUT; the registration problem still exists (even after using your rc)

-- 
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Sunday, September 26, 2004, 10:58:37 AM, you wrote:

DN On 26/09/2004 15:41 Vahan Yerkanian said the following:
 I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought
 several WellGate 3502A FXSes to play with till welltech guys fix the
 3504a's registration bug.

DN welltech still hasn't responded to my complaint to them regarding their SIP
DN registration process. in the interim however, i've patched asterisk 1.0 to
DN handle this wellgate special case. i'll be emailling the patch over to you
DN in a separate email. i run asterisk (0.9.0 and 1.0) on freebsd 4.10 too.

 alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's
 ports entries in the sip.conf, no voice is heard from both sides after
 the call is established.

DN i've seen the same. the wellgate's do not support the GSM codec and stock
DN asterisk does not do g723.1. you'd need to only use ulaw or alaw for them.
DN i've got this in my wellgate sip.conf entries:

DN disallow=all
DN allow=ulaw
DN allow=alaw


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Re[3]: [Asterisk-Users] Asterisk - WellGate 3502a : ulaw/alaw only?

2004-09-26 Thread Danny Zak
Hello Kiss,

well we didn't have this kind of errors .. YET :)



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Sunday, September 26, 2004, 3:56:39 PM, you wrote:

KK Hi,

KK Have some welltech devices myself, but after about one year of trials I
KK decided to throw them out ...
KK Even when you succeed in making it register it freezes after a random
KK amount of time, no matter what SW u use on it.

KK Regards,

KK Kiss Karoly

KK On Sun, 26 Sep 2004, Danny Zak wrote:

 Date: Sun, 26 Sep 2004 11:22:43 +0200
 From: Danny Zak [EMAIL PROTECTED]
 To: Dinesh Nair [EMAIL PROTECTED]
 Cc: Asterisk Users Mailing List - Non-Commercial Discussion
 [EMAIL PROTECTED]
 Subject: Re[2]: [Asterisk-Users] Asterisk - WellGate 3502a : ulaw/alaw
 only?

 Hello Dinesh,

 my welltech does support g729 g723.1 and g711u and a.

 It works like a charm here.

 BUT; the registration problem still exists (even after using your rc)

 --
 Best regards,
  Dannymailto:[EMAIL PROTECTED]

 belGOnet.com  a  Euro-pictures  division - internet solutions
 place princesse elisabeth 9/11   -   1030 Brussels  - Belgium
 Tel : +32-(0)2-215.67.65  -  Fax : +32-(0)2-215.66.65

 domains - hosting - hardware - VoiP - consultancy - backuping
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 belGOnet.com  accepts  no  liability for any damage resulting from the
 use and/or acceptation of the content of this email.


 Sunday, September 26, 2004, 10:58:37 AM, you wrote:

 DN On 26/09/2004 15:41 Vahan Yerkanian said the following:
  I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought
  several WellGate 3502A FXSes to play with till welltech guys fix the
  3504a's registration bug.

 DN welltech still hasn't responded to my complaint to them regarding their SIP
 DN registration process. in the interim however, i've patched asterisk 1.0 to
 DN handle this wellgate special case. i'll be emailling the patch over to you
 DN in a separate email. i run asterisk (0.9.0 and 1.0) on freebsd 4.10 too.

  alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's
  ports entries in the sip.conf, no voice is heard from both sides after
  the call is established.

 DN i've seen the same. the wellgate's do not support the GSM codec and stock
 DN asterisk does not do g723.1. you'd need to only use ulaw or alaw for them.
 DN i've got this in my wellgate sip.conf entries:

 DN disallow=all
 DN allow=ulaw
 DN allow=alaw


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[Asterisk-Users] spandsp

2004-09-26 Thread Danny Zak
Dear;

regarding the details that i found on the site about spandsp; is it
correct to assume the following ?

+ spandsp will only work with a card that is located in the * box?

therefore

will my welltech or any other voip fxo adapter support FAX ( i know it
does t.38)

--

My current situation is that i got a faxserver running in my old pbx
situation; i need to find a way to replace the whole pbx with a * box;
and need to fwd a certain MSN to my fax server, OR just have the * box
handle the incomming fax.

-- 
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 Danny  mailto:[EMAIL PROTECTED]

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[Asterisk-Users] Dialplan question

2004-09-26 Thread Danny Zak
Hello Asterisk,

  is it possible to make an extensions that write a call file
  (like a call back to the callerid)  in
  the outgoing directory WITHOUT using a perl AGI ?
  

-- 
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Re[2]: [Asterisk-Users] ISDN (point to point) questions

2004-09-25 Thread Danny Zak
Hello Bjoern,

thanks for this nice discussion;

we we dod have msn (4) although the telco company tells us that we
have pp isdn.  This seems to be a little bit strange to me... Is there
any way to crosscheck the isdn configuration ?

And what about the active or passive isdn cards ?
I just want to drive voice and fax over them.

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Saturday, September 25, 2004, 12:47:41 AM, you wrote:

BA Hi Danny,

BA dont mix ppp and p2p...


BA pp = point to point
BA pmp = point to multipoint
BA (both are ISDN connection configurations)

BA ppp = point to point protocol
BA a higher level protocol for data transmission etc.

BA Of which type your ISDN connection is, is usually easy to decide looking
BA at your phonenumbers

BA A PMP has MSNs (usually (in Germany) up to 10 with EuroISDN)
BA There may be up to (I'm not sure...) 8  devices connected to one PMP
BA ISDN connection.

BA A PP has one base number with DIDs (like 234567-0 234567-10 234567-11
BA etc). There may only be one device connected to the ISDN connection.
BA And with PP you can have several BRIs or PRIs 'sharing' the same number.

BA Greetings

BA Bjoern

BA Danny Zak schrieb:

 Hello;
 
 we are looking to replace our current PBX with a *-box; it is
 connected to ONE ppp isdn connection that is terminated by the NC.  We
 got on this box 4 msn's configured.
 
 currently we are working with pstn fxo's behind the PBX; it works but
 we can't use the CSID information behind it.  We want to migrate and
 keep the MSN's to decide routing in combination with the CID.
 
 That's why we want to replace all our phones (8) with voip ones (or by
 using a fxs's)
 
 Reading all information i assume the following
 
 - we need a ACTIVE ISDN card; these are 5/6 times more expensif that
 passive ones.
 I always assumed there was only PPP and PMP; but it seems -
 reading the specs of the av. isdn card - that there is also
 multilink PPP.  Will a multilink PPP also support a PPP; or is
 this just a other way to describe a PMP ?
 
 - how can i checke the number that is being dialed by the caller to
 reachh the * box (so one of the 4 msn's).  I have seen dialplans
 making use of the CALLERIDNUM; but what do i need to query for the
 called num ?
 


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Re[2]: [Asterisk-Users] Free G.729 ready for download

2004-09-24 Thread Danny Zak
Hello TELUX,

could anybody post something more about being legaly correct using
this codec and the corresponding royalty's.


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Friday, September 24, 2004, 9:45:29 PM, you wrote:

T Fantastic work!

T where do I get this from: /tmp/codec_g729/Makefile
T the patch command doesnt work for me, i think this is why.








T Daniel Pocock wrote:

 DISCLAIMER:  This code is free (I am not charging you to use it), but
 you might have to pay royalty fees to the G.729 patent holders for 
 using their algorithm.

 I finished this last Saturday and have had it on an Asterisk machine
 for 5 days without a crash, so I'm hoping that means it's safe to 
 release into the real world.

 This code has also been released on the -dev list.  As it is still 
 somewhat new,
 I would invite anyone with feedback to forward it to me personally or
 to the -dev list, as I don't monitor the -users list very often.

 http://www.readytechnology.co.uk/open/g729




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[Asterisk-Users] ISDN (point to point) questions

2004-09-24 Thread Danny Zak
Hello;

we are looking to replace our current PBX with a *-box; it is
connected to ONE ppp isdn connection that is terminated by the NC.  We
got on this box 4 msn's configured.

currently we are working with pstn fxo's behind the PBX; it works but
we can't use the CSID information behind it.  We want to migrate and
keep the MSN's to decide routing in combination with the CID.

That's why we want to replace all our phones (8) with voip ones (or by
using a fxs's)

Reading all information i assume the following

- we need a ACTIVE ISDN card; these are 5/6 times more expensif that
passive ones.
I always assumed there was only PPP and PMP; but it seems -
reading the specs of the av. isdn card - that there is also
multilink PPP.  Will a multilink PPP also support a PPP; or is
this just a other way to describe a PMP ?

- how can i checke the number that is being dialed by the caller to
reachh the * box (so one of the 4 msn's).  I have seen dialplans
making use of the CALLERIDNUM; but what do i need to query for the
called num ?

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Re[2]: [Asterisk-Users] Update: Welltech Wellgate 3504A registration problem

2004-09-21 Thread Danny Zak
Hello Leo,

3802 is doing the same (it is the fxo one)


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Tuesday, September 21, 2004, 9:07:00 AM, you wrote:



 essentially, by not keeping to the same Call-ID or tag, asterisk has
 no way of matching the wellgate's register with the past proxy auth
 packet (and thus the hashed md5 token).

LAB I've just gotten the box to register all 4-ports with an external SIP
LAB provider. The provider is running an old release of Broadsoft backend.
LAB Seems like Broadsoft supports this strange way of authentication.


 this has been reported to WellTech, but we've yet to get a response
 from them.

LAB They just sent me a new version of the 107a firmware. Have yet to check
LAB what's the difference from my running one.


 i've fashioned a patch which solves this behaviour and it works fine
 under asterisk 0.9.0 on freebsd and 1.0-RC1 on linux.

 i've got the patches for 0.9.0 and 1.0-RC1, and will be glad to email
 them if anyone wants them. the patch adds a new option under 
 [general], 'usenonce'. if usenonce=yes (default is no, normal asterisk
 behaviour), then asterisk will use the nonce tag as sent by the 
 wellgate to double check it's credentials and let it thru. this allows
 all 4 ports on the 3504A to be registered with a password.

LAB Any other gateways with the same problem?

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Re: [Asterisk-Users] Update: Welltech Wellgate 3504A registration problem

2004-09-20 Thread Danny Zak
Hello Leo,

you know anything about the 3802 ?

i also only got ONE port reged :(



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Monday, September 20, 2004, 12:15:49 PM, you wrote:

LAB Recently, there're some posts about registration problems with this
LAB gateway. These are my observations using the latest firmware version 107a.

LAB a) The gateway will register all 4 ports if you're not using password.
LAB b) If using password, the gateway will only register the 1st port correctly.

LAB For the record, the 2-port 3502 doesn't exhibit this behavior.

LAB FYI.



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Re[2]: [Asterisk-Users] H323 dialing makes Asterisk crash

2004-09-17 Thread Danny Zak
Hello Jeremy,

thanks for your remark..

this is what i get out of it ...


--

#0  0x41f57c50 in oh323_new (i=0x80f8f50, state=0,
host=0x449e5147 213.xxx.202.xxx) at chan_h323.c:625
625 chan_h323.c: No such file or directory.
in chan_h323.c


#0  0x41f57c50 in oh323_new (i=0x80f8f50, state=0,
host=0x449e5147 213.xxx.202.xxx) at chan_h323.c:625
#1  0x41f583b5 in oh323_request (type=0x449e5bdc H323, format=256,
data=0x449e598c) at chan_h323.c:793
#2  0x0805c330 in ast_request (type=0x449e5bdc H323, format=4, data=0x100)
at channel.c:1844
#3  0x40fd53d0 in dial_exec (chan=0x812c5c0, data=0x449e5bdc)
at app_dial.c:711
#4  0x08072136 in pbx_exec (c=0x812c5c0, app=0x41008ab0, data=0x449e7e2c,
newstack=1) at pbx.c:467
#5  0x08079e0e in pbx_extension_helper (c=0x812c5c0,
context=0x812c718 default, exten=0x0, priority=1,
callerid=0x80f8998 \admin\ portdanny, action=1090701488)
at pbx.c:1271
#6  0x0807406a in ast_pbx_run (c=0x4102c8b0) at pbx.c:1752
#7  0x0807a4d1 in pbx_thread (data=0x100) at pbx.c:1971
#8  0x4002a2b6 in start_thread () from /lib/tls/libpthread.so.0
--

i did recompile the h323 that is in the h323 dir; and everything goes
smooth there ...

i have worked in the past with h323 connection to a gatekeeper; but
that was just for testing ...

the h323.conf file contains gatekeeper=DISABLED

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Thursday, September 16, 2004, 11:12:52 PM, you wrote:

JM Danny Zak wrote:

 Hello Asterisk list;
 
 when i DIAL(H323/[EMAIL PROTECTED]) i get this strange error
 
 --
  -- Executing Dial(SIP/home-0953,
 H323/[EMAIL PROTECTED]|5|r) new stack
 backupns*CLI
 Disconnected from Asterisk server
 --
 
 Asterisk just goes down..



JM You are going to have to provide more debug information than that.


JM Make absolutely sure you have followed the instructions in 
JM asterisk/channels/h323/README



JM Jeremy McNamara

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[Asterisk-Users] H323 dialing makes Asterisk crash

2004-09-16 Thread Danny Zak
Hello Asterisk list;

when i DIAL(H323/[EMAIL PROTECTED]) i get this strange error

--
 -- Executing Dial(SIP/home-0953, H323/[EMAIL PROTECTED]|5|r) new stack
backupns*CLI
Disconnected from Asterisk server
--

Asterisk just goes down..




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Re[2]: [Asterisk-Users] sipphone dial out problems..

2004-09-13 Thread Danny Zak
Hello Marconi,

still the same

got the following when setting up call

-- Executing Dial(SIP/home-422a, SIP/[EMAIL PROTECTED]) in new
stack Sep 13 22:52:48 WARNING[376851]: chan_sip.c:590 __sip_xmit:
sip_xmit of 0x815e6e4 (len 677) to 198.65.166.131 returned -1: Invalid
argument -- Called [EMAIL PROTECTED] Sep 13 22:52:49

WARNING[114695]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x815e6e4 (len
677) to 198.65.166.131 returned -1: Invalid argument Sep 13 22:52:50

WARNING[114695]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x815e6e4 (len
677) to 198.65.166.131 returned -1: Invalid argument

strange .. since all other calls ( within the * )

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Sunday, September 12, 2004, 8:12:04 PM, you wrote:

MR On Sun, 12 Sep 2004 14:27:09 +0200, Danny Zak [EMAIL PROTECTED] wrote:
 hello;
 
 i got this each time when trying to dialout
 
 Sep 12 12:49:27 WARNING[114695]: chan_sip.c:590 __sip_xmit: sip_xmit
 of 0x811f88c (len 407) to 198.65.166.131 returned -1: Invalid argument
 
 sip.conf
 
 register = 1747668417x:[EMAIL PROTECTED]/1747668417x
 
 [sipphone]
 type=peer
 secret=
 username=1747668417x

MR Insert this:
MR fromuser = 1747668417x

 host=proxy01.sipphone.com
 nat=yes
 qualify=no
 host=dynamic
 reinvite=no
 canreinvite=no
 dtmfmode=inband
 
 extensions.conf
 
 exten = _1747.,1,SetCallerID,1747668417x
 exten = _1747.,2,Dial,SIP/${EXTEN:[EMAIL PROTECTED]

MR comment that, and put:

MR  exten = _1747.,1,Dial(SIP/[EMAIL PROTECTED])

 
 anyone a resolving issue on this ?
 
 --
 Best regards,
 Danny  mailto:[EMAIL PROTECTED]

MR Hope that solves your problem.

MR Regards,
MR Marconi.

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[Asterisk-Users] sipphone dial out problems..

2004-09-12 Thread Danny Zak
hello;

i got this each time when trying to dialout

Sep 12 12:49:27 WARNING[114695]: chan_sip.c:590 __sip_xmit: sip_xmit
of 0x811f88c (len 407) to 198.65.166.131 returned -1: Invalid argument

sip.conf

register = 1747668417x:[EMAIL PROTECTED]/1747668417x

[sipphone]
type=peer
secret=
username=1747668417x
host=proxy01.sipphone.com
nat=yes
qualify=no
host=dynamic
reinvite=no
canreinvite=no
dtmfmode=inband


extensions.conf

exten = _1747.,1,SetCallerID,1747668417x
exten = _1747.,2,Dial,SIP/${EXTEN:[EMAIL PROTECTED]


anyone a resolving issue on this ?


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Re[2]: [Asterisk-Users] WellGate 3504A with Asterisk SIP authenticationand config

2004-09-09 Thread Danny Zak
Hello Ariel's,

i got this back from welltech

  For the 38 unit
  It couldn't support only one account for the registeration as so far...

their reponse

We have the plan for this function...but it will be ok before the end of
Q4




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Thursday, September 9, 2004, 2:28:40 PM, you wrote:

AsH Olle E. Johansson wrote:
 Dinesh Nair wrote:


AsH This is just a quick note: The Wellgate's only register one user/password.
AsH If they have 2 ports they register them as one. They don't allow more then
AsH one password or username to be sent. They know about this problem but as far
AsH as I know they have not fixed it.  I tried for over 1 month to get them
AsH working but they had this one problem that they where not able to fix.


 hey * folk,

 am trying to configure a WellGate 3504A FXS SIP ATA
 (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've
 set up two SIP clients in sip.conf as follows:
 The Unathorized really suggests that the password is wrong for 1235.
 Check that you have the same secret on both asterisk and the welltech.

 Look at this packet. The welltech sends no username at all. How can
 we authorize that? I don't know if it's the missing account on
 1236 and 1237 that makes this happen.



 Sip read:
 REGISTER sip:192.168.0.200:5060 SIP/2.0
 Via: SIP/2.0/UDP
 192.168.0.202:5060;branch=z9hG4bK-413ef95c-6ea-7a64
 Max-Forwards: 70
 Supported: replaces
 User-Agent: FXS_GW (4asipfxs.107a)
 Contact: sip:@192.168.0.202:5060;expires=60
 From: sip:@192.168.0.200 ;tag=c0a800ca-13c4-413ef95c-6ea-3bf7
 To: sip:@192.168.0.200
 Call-ID: c0a800ca-13c4-0-334-1c34
 CSeq: 4 REGISTER
 Content-Length:0

 /Olle
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Re: [Asterisk-Users] WellGate 3504A with Asterisk SIP authentication and config

2004-09-08 Thread Danny Zak
Hello Dinesh,

i'm pretty sure that asterisk doesn't support multi logins for each line!


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Wednesday, September 8, 2004, 2:29:20 PM, you wrote:


DN hey * folk,

DN am trying to configure a WellGate 3504A FXS SIP ATA 
DN (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set
DN up two SIP clients in sip.conf as follows:

DN [general]
DN port = 5060 ; Port to bind to
DN bindaddr = 0.0.0.0  ; Address to bind SIP channel to
DN context = default   ; Default context for incoming calls

DN [1235]
DN host = dynamic
DN secret = somepass
DN context = default
DN type = friend
DN amaflags = billing
DN accountcode = SIPUSER
DN disallow=all
DN allow=ulaw
DN allow=alaw

DN [1234]
DN host = dynamic
DN secret = somepass
DN context = default
DN type = friend
DN amaflags = billing
DN accountcode = SIPUSER
DN disallow=all
DN allow=ulaw
DN allow=alaw

DN and on the wellgate i've set the following under SIP information:

DN Run Mode: Proxy
DN Primary Proxy IP Address: 192.168.0.200  this is the IP addy of
DN the asterisk server
DN Secondary Proxy IP Address: 
DN Outbound Proxy: 
DN Proxy port:  5060

DN Line 1 Number: 1234
DN Line 1 Account: 1234
DN Line 1 Password: somepass

DN Line 2 Number: 1235
DN Line 2 Account: 1235
DN Line 2 Password: somepass

DN Line 3 Number: 1236
DN Line 3 Account:
DN Line 3 Password:

DN Line 4 Number: 1237
DN Line 4 Account:
DN Line 4 Password:

DN SIP port:  5060
DN RTP Port:  16384
DN Expire:  60

DN however, only Line 1 from the wellgate succesfully authenticates with
DN asterisk. picking up the analog handset attached to line 1 (1234) works
DN fine with calls in and out working the way it should without any 
DN problems. Line 2 (and subsequently Lines 3  4) do not work at all. i've
DN captured the output of 'sip debug' and have attached it below. looking
DN at the output, it does seem that asterisk is rejecting the 
DN authentication for line 2 but it doesn't mention why. the obvious 
DN reasons (password mismatch et al) have been ruled out, of course.

DN any help would be much appreciated. i've read mailing list archive of
DN others being able to use the WellGate 3502 (which is just a 2xFXS port
DN version) with asterisk. however no mention of VoIP protocols was 
DN mentioned as the wellgates traditionally supported H.323 but with 
DN firmware upgrade has been able to support SIP.

DN (sip debug output begins)

DN Sip read:
DN REGISTER sip:192.168.0.200:5060 SIP/2.0
DN Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-0-35c-47a0
DN Max-Forwards: 70
DN Supported: replaces
DN User-Agent: FXS_GW (4asipfxs.107a)
DN Contact: sip:[EMAIL PROTECTED]:5060;expires=60
DN From: sip:[EMAIL PROTECTED] ;tag=c0a800ca-13c4-0-35c-48a3
DN To: sip:[EMAIL PROTECTED]
DN Call-ID: c0a800ca-13c4-0-334-1c34
DN CSeq: 1 REGISTER
DN Content-Length:0


DN 11 headers, 0 lines
DN Using latest request as basis request
DN Sending to 192.168.0.202 : 5060 (non-NAT)
DN Transmitting (no NAT):
DN SIP/2.0 100 Trying
DN Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-0-35c-47a0
DN From: sip:[EMAIL PROTECTED] ;tag=c0a800ca-13c4-0-35c-48a3
DN To: sip:[EMAIL PROTECTED];tag=as2b502f28
DN Call-ID: c0a800ca-13c4-0-334-1c34
DN CSeq: 1 REGISTER
DN User-Agent: Asterisk PBX
DN Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN Contact: sip:[EMAIL PROTECTED]
DN Content-Length: 0


DN   to 192.168.0.202:5060
DN Transmitting (no NAT):
DN SIP/2.0 407 Proxy Authentication Required
DN Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-0-35c-47a0
DN From: sip:[EMAIL PROTECTED] ;tag=c0a800ca-13c4-0-35c-48a3
DN To: sip:[EMAIL PROTECTED];tag=as2b502f28
DN Call-ID: c0a800ca-13c4-0-334-1c34
DN CSeq: 1 REGISTER
DN User-Agent: Asterisk PBX
DN Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
DN Contact: sip:[EMAIL PROTECTED]
DN Proxy-Authenticate: Digest realm=asterisk, nonce=00727441
DN Content-Length: 0


DN   to 192.168.0.202:5060


DN Sip read:
DN REGISTER sip:192.168.0.200:5060 SIP/2.0
DN Via: SIP/2.0/UDP
DN 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc
DN Max-Forwards: 70
DN Supported: replaces
DN User-Agent: FXS_GW (4asipfxs.107a)
DN Contact: sip:[EMAIL PROTECTED]:5060;expires=60
DN From: sip:[EMAIL PROTECTED]
DN ;tag=c0a800ca-13c4-413ef95b-488-4bff
DN To: 

Re[2]: [Asterisk-Users] Festival TTS mbrola ?

2004-09-02 Thread Danny Zak
Hello Eric,

thanks for the information

i have everything installed .. but can't seem to get the voice to get
into asterisk

i tried

System(/usr/local/play_ola /tmp/dino.ola)

as a test .. but it times out ..

any solutions ?


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Thursday, September 2, 2004, 9:44:00 AM, you wrote:

EB I have tested positively mbrola with lliaphon for the french language.
EB I'm using it within an agi script.

EB for lliaphon go :
EB http://www.culte.org/projets/biglux/install/lao/lliaphon.shtml

EB for other languages try :
EB http://tcts.fpms.ac.be/synthesis/mbrola/mbrtts.html

EB Regards
EB Eric
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[Asterisk-Users] Festival TTS mbrola ?

2004-09-01 Thread Danny Zak
Hello Asterisk,

  noticed that the mbrola really adds a extra dimension to tts; anyone
  got any experience with running this together with the festival and
  * ?

-- 
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 Danny  mailto:[EMAIL PROTECTED]

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[Asterisk-Users] good Dutch TTS ?

2004-08-31 Thread Danny Zak
hi;

anyone can recommend a good TTS for the dutch language compat in
linux?

-- 
Best regards,
 Danny  mailto:[EMAIL PROTECTED]

belGOnet.com  a  Euro-pictures  division - internet solutions
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[Asterisk-Users] Read and dtmf ?

2004-08-25 Thread Danny Zak
Hello asterisk-users,

  i got a simple read configured as an extensions command.

  it does not detect the dtmf digits all god

  0123456789
  gives something like

  0223455789


  what could this be the issue ?

  i'm using g711u law with inband.

  i tried to use outband but i got hardware like FXO's that don't
  support outband
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 Danny  mailto:[EMAIL PROTECTED]

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[Asterisk-Users] welltech fxo and *

2004-08-21 Thread Danny Zak
hello everybody ,

  i got a test setup with

  lanphone 101 connected to *

  and a welltech fxo 3802 also to *


  the extensions are configure so that i can dial from the lanphone to
  the fxo.

  although once on the FXO and having the dialtone, no of my dtmf
  dialing is being processed on the FXO.  It keeps giving me the
  dialtone constatly.

  Everything is configured as outband DTMF (we tried also doing inband
  but it was doing the same)

  We tried to call the echo test application to check if the # would
  close the connection, and it does.  So dtmf from lanphone to * seems
  to work OK.

  We also tried to configure a extension like

  dial(SIP/FXO,50,D(0022156765))

  it does connect to the fxo; but the dtmf isn't being processed...
  (keeps giving me the dialtone)
  
  Anybody got a similar problem like this, and a prob. solution ?

  the sip.conf for each unit looks like

[home]
disallow=all
allow=ulaw
type=friend
secret=
username=home
host=dynamic
context=default
nat=yes
dtmfmode=rfc2833



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Best regards,
 Danny  mailto:[EMAIL PROTECTED]

belGOnet.com  a  Euro-pictures  division - internet solutions
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